[1] In an echo canceller utilized in order to realize full duplex in a hands-free telephone set, a television conference, and so forth, an adaptive filter coefficient has been conventionally updated using an alternate voice, to estimate an impulse response of an echo path. As this estimation algorithm, the Normalized Least Mean Square Method (the NLMS Method) having a relatively small amount of operation has been frequently used.
FIG. 1 illustrates an example of a conventional echo canceller 1. The echo canceller 1 is constructed as a hybrid (line) echo canceller for canceling a hybrid echo, and updates an adaptive filter coefficient by the NLMS Method.
Referring to FIGS. 1 and 2, description is made of the operation of the echo canceller 1.
An echo signal y, an echo replica signal (a pseudo echo signal) Y, an adaptive filter coefficient, a value x in a reference input signal buffer, a cancellation error signal e, and so forth are initialized (step 1).
A microphone input signal which is digitized by an analog-to-digital (A/D) converter 2 is sampled, and is inputted to a reference input signal buffer 3 as a reference input signal X (step 3).
An adaptive filter coefficient Pj (i) in an adaptive filter 4 and the reference input signal X, i.e., a value xj (i) in the reference input signal buffer 3 at time j are multiplied and accumulated by a multiply and accumulate unit 5, thereby to generate an echo replica signal Yj at the time j, as expressed by the following equation (1) (step 5):                     Yj        =                              ∑                          i              =              1                        N                    ⁢                                           ⁢                                    Pj              ⁡                              (                i                )                                      ·                          xj              ⁡                              (                i                )                                                                        (        1        )            
A subtractor 7 subtracts the echo replica signal Yj from an echo signal yj outputted from an A/D converter 6, thereby to calculate a cancellation error signal ej at the time j, as expressed by the following equation (2), (step 7):ej=yj−Yj  (2)
Thereafter, an adaptive filter coefficient Pj+1 (i) is updated by a coefficient updating unit 8 on the basis of the cancellation error signal ej the adaptive filter coefficient Pj (i), and the reference input signal X (=xj (i)) at the time j as expressed by the following equation (3) (step 9). That is, an adaptive filter coefficient at time (j+1) is found:                               Pj          +                      1            ⁢                          (              i              )                                      =                              Pj            ⁡                          (              i              )                                +                                    μ              ⁢                                                           ⁢                              ej                ·                                  xj                  ⁡                                      (                    i                    )                                                                                                      ∑                                  i                  =                  1                                N                            ⁢                                                           ⁢                                                xj                  ⁡                                      (                    i                    )                                                  2                                                                        (        3        )            
The processing at the foregoing steps 3 to 9 is repeated, so that the adaptive filter coefficient is updated.
In the above-mentioned equations (1) to (3), yj denotes an echo signal at the time j, Yj denotes an echo replica signal at the time j, Pj (i) denotes a coefficient of the i-th delay element in the adaptive filter at the time j, xj (i) denotes the i-th value of the reference input signal buffer at the time j ej denotes a cancellation error signal at the time j, N denotes the number of taps in the adaptive filter, and μ denotes a relaxation coefficient called a step gain. The same is true for embodiments of the invention, described later.
In the above-mentioned echo canceller 1, the adaptive filter coefficient is updated along the vector of the reference input signal X(=xj (i)), as expressed by the foregoing equation (3). However, the reference input signal is a voice signal which is high in auto-correlation, so that the learning speed is reduced. The vector of the reference input signal is a value, grasped as a vector, in the reference input signal buffer 3.
As indicated by the second term on the right side of the equation (3), μej·xj(i) is normalized by the norm=Σxj(i)2 of the vector of the reference signal. Accordingly, the learning precision in a frequency area having a small number of frequency components is degraded by the deviation in frequency components of a voice signal.
When an impulse response of an echo path is estimated using an alternate voice, as in the echo canceller 1, therefore, frequency components of an input signal deviate. Even when learning is sufficiently performed, therefore, howling may, in some cases, occur.
The frequency characteristics of a voice signal are generally as shown in FIGS. 3a and 3b. That is, voiced speech (a voiced sound) has the property of decreasing in level by 6 dB when the frequency thereof increases by one octave (−6 dB/oct), as shown in FIG. 3a. Unvoiced speech (a voiceless sound) has the property of increasing in level by 6 dB when the frequency thereof increases by one octave (+6 dB/oct), as shown in FIG. 3b. 
In order to improve the learning speed and the learning precision of an echo canceller using the NLMS Method, therefore, it is considered that the deviation in frequency characteristics of a signal used for learning is decreased.
Therefore, it is considered that a whitening filter 9 for decreasing the deviation in frequency characteristics of a voice signal is inserted between the A/D converter 2 and a digital-to-analog (D/A) converter 9, as in an echo canceller la shown in FIG. 4. In the echo canceller 1a, however, an output signal of the whitening filter 9 is outputted toward the hybrid side through the D/A converter 9, so that the sound quality of the output signal is degraded.
An object of the present invention is to provide echo canceling means and an echo canceller capable of improving the learning speed and the learning precision without degrading the sound quality of an output signal.
[2] In a voice switch used for a hands-free speech communication system, a signal in a communication path through which the voice of one of a near-end speaker and a far-end speaker who is talking by phone passes is passed, while a signal in a communication path through which the voice of the speaker who is talking by phone does not pass is attenuated by an attenuator, thereby preventing an echo from being returned to the communication path through which the voice of the speaker who is talking by phone does not pass.
In the voice switch, when one of the near-end speaker and the far-end speaker starts to talk by phone from a state where both the speakers do not talk by phone, the beginning or the ending, for example, of a word may, in some cases, be cut.
An object of the present invention is to provide a voice switch capable of preventing the beginning or the ending of a word from being cut, for example, when one of a near-end speaker and a far-end speaker starts to talk by phone from a state where both the speakers do not talk by phone as well as capable of carrying out more natural telephone conversation which hardly has a switching feeling.